Currently, VoIP (Voice over Internet Protocol) over a mobile broadband network such as the High Speed Packet Access (HSPA), Long Term Evolution (LTE), Enhanced Data Rates for GSM Evolution (EDGE) or EDGE Continued Evolution has been given much attention. The main reason for choosing these mobile broadband networks, over other radio bearer realizations is flexibility. This flexibility promises easy introduction of new media streams as a complement to the voice stream.
However, flexible scheduling in these mobile broadband networks raises concern that the performance of the VoIP service will be negatively influenced by significant amounts of jitter. Jitter can be viewed as a variation of packet transit delay caused by queuing contention, and serialization effects on the path through the network.
One way to handle the jitter is to introduce a jitter buffer. The jitter buffer helps to remove the effects of jitter when decoding the voice stream, buffering each arriving VoIP packet for a short interval before playing it out. The jitter buffer can be considered as a time window with one side (the early side) aligned with a minimum possible delay and the other side (the late side) representing a maximum possible delay before a packet would be discarded. A concern of the jitter buffer is that the end-to-end delay of the VoIP service will be significantly longer than for a commercial CS (circuit switched) telephony system.
In Voice over HSPA systems and Voice over LTE systems, the main part of the jitter is introduced by the packet scheduler and the fast retransmission mechanism between the base station and the mobile terminals called the H-ARQ. The scheduler, depending on implementation and load conditions, can decide to send one packet to the receiver as soon as it arrives to the scheduler or wait and possibly send several packets in one transmission to the receiver which introduces jitter. Due to the real-time characteristics of VoIP, the scheduler for VoIP should not wait to long. Typically, a maximum scheduling delay threshold is implemented in the scheduler and packets older than the maximum threshold are discarded. The H-ARQ retransmits the packets (one or several) until either the receiving entity successfully can decode the received information or until the maximum retransmission delay threshold is passed, leading to the packet being discarded.
Voice over EDGE has also gained interest since many operators have a mixture of WCDMA (Wide Band Code Division Multiple Access) and GSM (Global System for Mobile communications) networks. Like GSM, EDGE has time-sharing and scheduled access. In the EDGE Continued Evolution, non-persistent scheduling is a possibility. As for the H-ARQ mechanisms in the HSPA and LTE, this implies that a VoIP packet will be retransmitted only a few times at most before it is discarded. This increases the probability of transmission, but also raises concerns that the performance of the VoIP over EDGE will be negatively influenced by significant amounts of jitter.
To illustrate this problem, a VoIP over HSPA example is shown in FIG. 7. FIG. 7 presents a calculation of an end-to-end delay in a VoIP over HSPA network. From FIG. 7, a conclusion can be drawn that the end-to-end delay may vary between 134 ms and 229 ms in this particular instance. Thus, the jitter buffer 220 should be 229−134=95 ms long. This corresponds to the difference between the minimum scheduling delay of the DL (downlink) transmission, i.e. from the network to the mobile UE (user equipment), and the maximum scheduling delay of the DL. However, in conventional systems, the knowledge about the maximum scheduling threshold is not known by the UE. The mobile UE is also referred to as a mobile wireless terminal.